asterisk

Commit Graph

Author	SHA1	Message	Date
Richard Mudgett	a0f8821749	Add dialtone_detect option for analog incoming calls. For analog lines, enables Asterisk to use dialtone detection per channel if an incoming call was hung up before it was answered. If dialtone is detected, the call is hung up. no: Disabled. (Default) yes: Look for dialtone for 10000 ms after answer. <number>: Look for dialtone for the specified number of ms after answer. always: Look for dialtone for the entire call. Dialtone may return if the far end hangs up first. dialtone_detect=yes dialtone_detect=5000 dialtone_detect=always (closes issue ASTERISK-19316) Reported by: Jeremy Pepper Patch by: Jeremy Pepper Tested by: rmudgett,Jeremy Pepper Review: https://reviewboard.asterisk.org/r/1737/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358344 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	62aae50142	Add IPv6 support to FastAGI. Review: https://reviewboard.asterisk.org/r/1774/ Reviewed by: Simon Perreault, Mark Michelson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357610 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	299dd5d4fc	Adds an option to sip.conf that prevents diversion headers from being added. send_diversion=no will prevent Diversion headers from being added to SIP requests. This doesn't prevent Diversion from being added with dialplan such as with SIPAddHeader. (closes issue ASTERISK-16862) Reported by: rsw686 Review: https://reviewboard.asterisk.org/r/1769/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356987 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	a78b0af5ea	Re-commit the verbose branch. This change permits each verbose destination (consoles, logger) to have its own concept of what the verbosity level is. The big feature here is that the logger will now be able to capture a particular verbosity level without condemning each console to need to suffer that level of verbosity. Additionally, a stray 'core set verbose' will no longer change what will go to the log. Review: https://reviewboard.asterisk.org/r/1599/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355413 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	33322e38f0	res_agi: Add AGIEXITONHANGUP variable. This patch adds a variable AGIEXITONHANGUP for res_agi. If this variable is set to "yes" on a channel, AGI() will exit immediately once a channel hangup has been detected. This was the behavior of AGI() in Asterisk 1.4 and earlier and is still desired by some people. Review: https://reviewboard.asterisk.org/r/1734/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355102 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	e5c51ee44c	Add auto_force_rport and auto_comedia NAT options This patch adds the auto_force_rport and auto_comedia NAT options. It also converts the nat= setting to a list of comma-separated combinable options: no, force_rport, comedia, auto_force_rport, and auto_comedia. nat=yes remains as an undocumented option equal to "force_rport,comedia". The first instance of 'yes' or 'no' in the list stops parsing and overrides any previously set options. If an auto_* option is specified with its non-auto_ counterpart, the auto setting takes precedence. This patch builds upon the patch posted to ASTERISK-17860 by JIRA user pedro-garcia. (closes issue ASTERISK-17860) Review: https://reviewboard.asterisk.org/r/1698/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354597 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	3342183016	Add callbackextension matching & realtime callbackextensions This patch is based on the one by David Vossel, developer extrodinaire, at https://reviewboard.asterisk.org/r/344/. If multiple peers are defined with the same host/port, but differing callbackextensions, it chooses the peer with the matching callbackextension. Since callbackextension creates an outbound registration with the callbackextension as the Contact address, matching an incoming request by that (in addition to the host/port) makes a lot of sense. This patch also adds support for callbackextension to realtime by querying all peers with callbackextensions on reload and adding registrations for them. (closes issue ASTERISK-13456) Review: https://reviewboard.asterisk.org/r/344/ Review: https://reviewboard.asterisk.org/r/1717/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354458 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	0adeb88318	Add CHANGES documentation for the "pri set debug" bitmask change (related to ASTERISK-17159) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354395 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	055a19e128	Replace res_ais with a new module, res_corosync. This patch removes res_ais and introduces a new module, res_corosync. The OpenAIS project is deprecated and is now just a wrapper around Corosync. This module provides the same functionality using the same core infrastructure, but without the use of the deprecated components. Technically res_ais could have been used with an AIS implementation other than OpenAIS, but that is the only one I know of that was ever used. Review: https://reviewboard.asterisk.org/r/1700/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354046 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	92ef8a6fe1	Address OpenSSL initialization issues when using third-party libraries. When Asterisk is used with various third-party libraries (CURL, PostgresSQL, many others) that have the ability themselves to use OpenSSL, it is possible for conflicts to arise in how the OpenSSL libraries are initialized and shutdown. This patch addresses these conflicts by 'wrapping' the important functions from the OpenSSL libraries in a new shared library that is part of Asterisk itself, and is loaded in such a way as to ensure that all calls to these functions will be dispatched through the Asterisk wrapper functions, not the native functions. This new library is optional, but enabled by default. See the CHANGES file for documentation on how to disable it. Along the way, this patch also makes a few other minor changes: * Changes MODULES_DIR to ASTMODDIR throughout the build system, in order to more closely match what is used during run-time configuration. * Corrects some errors in the configure script where AC_CHECK_TOOLS was used instead of AC_PATH_PROG. * Adds a new variable for linker flags in the build system (DYLINK), used for producing true shared libraries (as opposed to the dynamically loadable modules that the build system produces for 'regular' Asterisk modules). * Moves the Makefile bits that handle installation and uninstallation of the main Asterisk binary into main/Makefile from the top-level Makefile. * Moves a couple of useful preprocessor macros from optional_api.h to asterisk.h. Review: https://reviewboard.asterisk.org/r/1006/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353317 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	cbe57b11cb	Fixes for sending SIP MESSAGE outside of calls. * Fix authenticate MESSAGE losing custom headers added by the MESSAGE_DATA function in the authorization attempt. * Pass up better From header contents for SIP to use. Now is in the "display-name" <URI> format expected by MessageSend. (Note that this is a behavior change that could concievably affect some people.) * Block user from adding standard headers that are added automatically. (To, From,...) * Allow the user to override the Content-Type header contents sent by MessageSend. * Decrement Max-Forwards header if the user transferred it from an incoming message. * Expand SIP short header names so the dialplan and other code only has to deal with the full names. * Documents what SIP expects in the MessageSend(from) parameter. (closes issue ASTERISK-18992) Reported by: Yuri (closes issue ASTERISK-18917) Reported by: Shaun Clark Review: https://reviewboard.asterisk.org/r/1683/ ........ Merged revisions 352520 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352538 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	de09749470	Add an announcement option to music-on-hold - plays sound when put on hold/between songs This is a feature patch which allows an 'announcement' option to be specified in musiconhold.conf which should be set to the name of a sound. If a valid sound is specified for this option, then it will be played on that music on hold class whenever a channel bound to that class is put on hold as well as when Asterisk is able to detect that a song has ended before starting the next song (excludes external players). (closes ASTERISK-18977) Reported by: Timo Teräs Patches: asterisk-moh-announcement.diff uploaded by Timo Teräs (license 5409) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352134 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	a1bef6041d	Adds the ability to stop specific mixmonitors by using unique IDs set at monitor launch. MixMonitor receives a new option i(channel_variable) which stores the unique id at said variable. StopMixMonitor now accepts ID as an optional argument, which if included will make StopMixMonitor specifically target the mixmonitor on that particular channel. CLI commands and AMI actions have been ammended to work with the IDs as well. In addition, monitors across a channel can now be listed be listed via CLI command "mixmonitor list <channel>" which will display all of the mixmonitors active on that channel along with the files they each have open. Created by Sergio González Martín. (closes issue ASTERISK-19096) Reported by: Sergio González Martín Review: https://reviewboard.asterisk.org/r/1643/ Review: https://reviewboard.asterisk.org/r/1682/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352093 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	778fa4abaf	Various parking improvements. * Adds per-parking lot options comebackcontext and comebackdialtime * Makes comebacktoorigin settable per parking lot * Sets a PARKER channel variable when comebacktoorigin is disabled (closes issue ASTERISK-16643) Reported by: Mitch Sharp (bluecrow76) Patches: asterisk-1.6.2.17.2-park-features-comebackcontext-consolidated-v3.diff by Mitch Sharp (bluecrow76) license 5231 with updates by me. Review: https://reviewboard.asterisk.org/r/1674 Review: https://reviewboard.asterisk.org/r/963 Reviewed by Richard Mudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351913 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	ee4cf38a27	Adds pjmedia probation concepts to res_rtp_asterisk's learning mode. In order to better handle RTP sources with strictrtp enabled (which is now default in 10) using the learning mode to figure out new sources when they change is handled by checking for a number of consecutive (by sequence number) packets received to an rtp struct based on a new configurable value called 'probation'. Also, during learning mode instead of liberally accepting all packets received, we now reject packets until a clear source has been determined. Review: https://reviewboard.asterisk.org/r/1663/ ........ Merged revisions 351287 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351289 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351290 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	c60d15222c	Add ABS() absolute value function to the expression parser. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351079 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	edf466012f	Make FollowMe optionally update connected line information when the accepting endpoint is bridged. Like Dial and Queue, FollowMe needs to deal with AST_CONTROL_CONNECTED_LINE information so when the parties are initially bridged, the connected line information will be correct. * Added the 'I' option just like the app_dial and app_queue 'I' option. * Made 'N' option ignored if the call is already answered. (closes issue ASTERISK-18969) Reported by: rmudgett Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1656/ ........ Merged revisions 350364 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350415 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350416 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	b0243fb57c	Allow overriding of IMAP server settings on a user by user basis This patch allows the imapserver, imapport, and imapflags settings to be overridden for any voicemail user. It also documents the settings in the sample voicemail.conf file, and updates the voicemail schema to allow storage of those columns. (closes issue ASTERISK-16489) Reporter: Hubert Mickael Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1614/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349106 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	03596bcb47	chan_sip autocreatepeer=persist option for auto-created peers to survive reload This patch moves destruction of sip peers to immediately after the general section of sip.conf is read so that autocreatepeer setting can be read before deletion of peers. If autocreatepeer=persist at reload, then peers created by the autocreatepeer setting will be skipped when purging the current SIP peer list. (closes ASTERISK-16508) Reported by: Kirill Katsnelson Patches: 017797-kkm-persist-autopeers-1.8.patch uploaded by Kirill Katsnelson (license 5845) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349097 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	1b0741c7db	Voicemail with the saycid option will now play a caller's name based on cid if available. In order to check the availability of the caller's name, app_voicemail will check for an audio file in <astspooldir>/recordings/callerids/ This change sets a precedent for where to put recordings of names. Currently the idea is that recordings here could also be used for applications like confbridge and meetme to find recorded names in this folder from callerid (when another recording isn't available) (closes issue ASTERISK-18565) Reporter: Russell Brown Patches: r uploaded by Russel Brown (license 6182) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348416 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	9057aa20b6	Backed out core changes from r346391 During testing, it was discovered that there were a number of side effects introduced by r346391 and subsequent check-ins related to it (r346429, r346617, and r346655). This included the /main/stdtime/ test 'hanging', as well as the remote console option failing to receive the appropriate output after a period of time. I only backed out the changes to main/ and utils/, as this was adequate to reverse the behavior experienced. (issue ASTERISK-18974) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347997 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	518ccb6706	Remove autojump extensions from SayUnixTime, make an option to perform automatic jumps. When a caller sends DTMF while the SayUnixTime application is saying the time, The call would jump to the next extension much like it does during Background(). This patch adds option 'j' to SayUnixTime which when used employs the old behavior. Also, this patch allows arguments to sayunixtime to not be used as empty strings in the case of something like 'sayunixtime(,,,j)' or 'sayunixtime(,,pattern). (closes issue ASTERISK-16675) Reported by: jlpedrosa Patches: patch_SayUnixTime_noJump.patch uploaded by jlpedrosa (license 5959) Review: https://reviewboard.asterisk.org/r/956/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347866 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	fd64bb66f9	Add VM_INFO() dialplan function to gather information about a mailbox. Deprecates MAILBOX_EXISTS. Provides count, email, exists, fullname, language, locale, pager, password, tz. (closes issue ASTERISK-18634) Patch by: Kris Shaw Review: https://reviewboard.asterisk.org/r/1568 Reviewed by: Walter Doekes git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347192 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	77b670c4ab	Allow each logging destination and console to have its own notion of the verbosity level. Review: https://reviewboard.asterisk.org/r/1599 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346391 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	32d0faac9c	Default to nat=yes; warn when nat in general and peer differ It is possible to enumerate SIP usernames when the general and user/peer nat settings differ in whether to respond to the port a request is sent from or the port listed for responses in the Via header. In 1.4 and 1.6.2, this would mean if one setting was nat=yes or nat=route and the other was either nat=no or nat=never. In 1.8 and 10, this would mean when one was nat=force_rport and the other was nat=no. In order to address this problem, it was decided to switch the default behavior to nat=yes/force_rport as it is the most commonly used option and to strongly discourage setting nat per-peer/user when at all possible. For more discussion of the issue, please see: http://lists.digium.com/pipermail/asterisk-dev/2011-November/052191.html (closes issue ASTERISK-18862) Review: https://reviewboard.asterisk.org/r/1591/ ........ Merged revisions 345776 from http://svn.asterisk.org/svn/asterisk/branches/1.4 ........ Merged revisions 345800 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........ Merged revisions 345828 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345830 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345831 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	298d015828	Add #tryinclude statement This provides the same functionality as #include however an asterisk module will still load if the filename does not exist. Review: https://reviewboard.asterisk.org/r/1476/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345735 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	279873e8eb	Add admin toggle mute all and participant count menu options to app_confbridge This patch adds two new menu features to app_confbridge, admin_toggle_menu_ participants and participant_count. The admin action will globally mute / unmute all non-admin participants on a converence, while the participant count simply exposes the existing participant count function to the conference bridge menu. This also adds configuration options to change the sound played when the conference is globally muted / unmuted, as well as the necessary config hooks to place these functions in the DTMF menus. (closes issue ASTERISK-18204) Reported by: Kevin Reeves Tested by: Matt Jordan Patches: app_confbridge.c.patch.txt, conf_config_parser.c.patch.txt, confbridge.h.patch.txt uploaded by Kevin Reeves (license 6281) Review: https://reviewboard.asterisk.org/r/1518/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345560 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	02f886b5a2	Allow built in variables to be used with dynamic weights. You can now use the built in variables , , and within a dynamic weight. For example, this could be useful when you want to pass requested lookup number to the SHELL() function which could be used to execute a script to dynamically set the weight of the result. (Closes issue ASTERISK-13657) Reported by: Joel Vandal Tested by: Leif Madsen, Russell Bryant Patches: asterisk-1.6-dundi-varhead.patch uploaded by Joel Vandal (License #5374) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343693 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	71b7df16bf	Merged revisions 341580 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r341580 \| irroot \| 2011-10-20 19:13:23 +0200 (Thu, 20 Oct 2011) \| 15 lines Add option to check state when state is unknown r341486 reverts r325483 this is a rework of the patch. optimize to minimize load. add option check_state_unknown to control whether a member with unknown device state is checked there is a small % chance that calls will be sent to the member when they on a call. app_queue will see a device with unknown state as available and does not try verify the state without this option enabled. Review: https://reviewboard.asterisk.org/r/1535/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341583 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	a79c41ee66	Add information about limitations of new codec support in channel drivers. (issue ASTERISK-18680) ........ Merged revisions 341094 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341096 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	b698038995	Add generic faxdetect framehook to res_fax Added func FAXOPT(faxdetect)=yes,cng,t38[,timeout]/no to enable dialplan faxdetect allowing more flexibility. as soon as a fax tone is detected the framehook is removed. there is a penalty involved in running this framehook on non G711 channels as they will be transcoded. CNG tone is suppresed using the SQUELCH flag to allow WaitForNoise to be run on the channel to detect Voice. (Closes issue ASTERISK-18569) Reported by: Myself Reviewed by: Matthew Nicholson, Kevin Fleming Review: https://reviewboard.asterisk.org/r/1116/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339465 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	eeefca7f91	Generate error message when AMI action originate extension doesn't exist Review: https://reviewboard.asterisk.org/r/1445/ Is this a bug or a new feature? No responses on Asterisk-dev so I'm committing to trunk only. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339206 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	7b26fde494	Merged revisions 338997 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r338997 \| irroot \| 2011-10-03 16:38:25 +0200 (Mon, 03 Oct 2011) \| 1 line Documentation noting the extension of CHANNEL() for chan_ooh323 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338998 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	383b073966	Add CLI command "cdr show pgsql status" based on "cdr mysql status" Review: https://reviewboard.asterisk.org/r/923/ Thanks all for the code reviews and feedback. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338415 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	659edb7b8f	Update CHANGES to reflect autopausebusy not being in Asterisk 10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338188 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	0ab04b53b5	Add autopausebusy and autopauseunavail queue options Make it possible to autopause on a busy or unavailable response from a device. (closes issue ASTERISK-16112) Reported by: jlpedrosa Patches: autopausebusy.txt by twilson Review: https://reviewboard.asterisk.org/r/1399/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338187 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	5982bdcb7c	Merged revisions 337595,337597 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r337595 \| jrose \| 2011-09-22 10:35:50 -0500 (Thu, 22 Sep 2011) \| 12 lines Generate Security events in chan_sip using new Security Events Framework Security Events Framework was added in 1.8 and support was added for AMI to generate events at that time. This patch adds support for chan_sip to generate security events. (closes issue ASTERISK-18264) Reported by: Michael L. Young Patches: security_events_chan_sip_v4.patch (license #5026) by Michael L. Young Review: https://reviewboard.asterisk.org/r/1362/ ........ r337597 \| jrose \| 2011-09-22 10:47:05 -0500 (Thu, 22 Sep 2011) \| 10 lines Forgot to svn add new files to r337595 Part of Generating security events for chan_sip (issue ASTERISK-18264) Reported by: Michael L. Young Patches: security_events_chan_sip_v4.patch (License #5026) by Michael L. Young Reviewboard: https://reviewboard.asterisk.org/r/1362/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337600 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	8f10934c18	Merged revisions 337261 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r337261 \| irroot \| 2011-09-21 12:42:06 +0200 (Wed, 21 Sep 2011) \| 10 lines Adds a timeout argument to app_originate the default is 30s this will be used if the timout supplied is invalid or no timeout is supplied. Contributed by: jacco (thank you for the work) Review: https://reviewboard.asterisk.org/r/1310/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337262 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	7b08b2cf53	Merged revisions 337219 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r337219 \| oej \| 2011-09-21 11:32:50 +0200 (Ons, 21 Sep 2011) \| 13 lines Make ast_pbx_run() not default to s@default if extension is not found Review: https://reviewboard.asterisk.org/r/1446/ This is a bug - or architecture mistake - that has been in Asterisk for a very long time. It was exposed by the AMI originate action and possibly some other applications. Most channel drivers checks if an extension exists BEFORE starting a pbx on an inbound call, so most calls will not depend on this issue. Thanks everyone involved in the review and on IRC and the mailing list for a quick review and all the feedback. (closes issue ASTERISK-18578) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337220 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	2ae7ae00c8	Merged revisions 337178 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r337178 \| oej \| 2011-09-21 10:51:41 +0200 (Ons, 21 Sep 2011) \| 14 lines Change strictrtp option to default to yes in the RTP module Suggested by Kapejod on Facebook Review: https://reviewboard.asterisk.org/r/1448/ (closes issue ASTERISK-18587) Thanks for quick feedback to kpfleming and Tilghman --Denna och nedanstående rader kommer inte med i loggmeddelandet-- M CHANGES M configs/rtp.conf.sample M res/res_rtp_asterisk.c ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337179 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	8493c46308	Merged revisions 336936 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r336936 \| irroot \| 2011-09-20 18:51:59 +0200 (Tue, 20 Sep 2011) \| 14 lines Allow Setting Auth Tag Bit length Based on invite or config option Update the SIP SRTP API to allow use of 32 or 80 bit taglen. Curently only 80 bit is supported. The outgoing invite will use the taglen of the incoming invite preventing one-way audio. (Closes issue ASTERISK-17895) Review: https://reviewboard.asterisk.org/r/1173/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336937 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	73424f128e	Merged revisions 336042 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r336042 \| oej \| 2011-09-15 14:46:38 +0200 (Tor, 15 Sep 2011) \| 12 lines Meetme: Introducing a new option "k" to kill a conference if there's only a single member left. When using Meetme as a modular call bridge from third party applications, it's handy to make it behave like a normal call bridge. When the second to last person exists, the last person will be kicked out of the conference when this option is enabled. (closes issue ASTERISK-18234) Review: https://reviewboard.asterisk.org/r/1376/ Patch by oej, sponsored by ClearIT, Solna, Sweden ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336043 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	8017b65bb9	Merged revisions 335014 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r335014 \| irroot \| 2011-09-09 09:23:53 +0200 (Fri, 09 Sep 2011) \| 9 lines Move code for VALID_EXTEN from app_readexten to func_dialplan Mark VALID_EXTEN deprecated. Review: https://reviewboard.asterisk.org/r/1396/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335015 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alec L Davis	5ad57732f5	Merged revisions 334621 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r334621 \| alecdavis \| 2011-09-07 20:14:50 +1200 (Wed, 07 Sep 2011) \| 9 lines Merged revisions 334620 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334620 \| alecdavis \| 2011-09-07 20:12:49 +1200 (Wed, 07 Sep 2011) \| 2 lines peroid typo ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334623 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	f03bccdb4d	Implement the '!' negation element to negate codecs directly in the allow keyword. This permits the list of codecs to be specified in one configuration line, instead of two or more, generally with the aim of either allowing all codecs with the exception of a few or disallowing most but permitting a few. Review: https://reviewboard.asterisk.org/r/1411/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334574 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	39ac2e639f	Merged revisions 334514 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r334514 \| pabelanger \| 2011-09-06 11:47:59 -0400 (Tue, 06 Sep 2011) \| 6 lines authdebug is now disabled by default To enable this functionaility again set authdebug = yes in iax.conf Review: https://reviewboard.asterisk.org/r/1414/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334517 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	9d2af5071b	Merged revisions 333681 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r333681 \| twilson \| 2011-08-29 12:28:59 -0500 (Mon, 29 Aug 2011) \| 7 lines Use realtime text when it is negotiated This patch make use of wirte_text() realtime text instead of send_text() if T.140 is in native formats. ASTERISK-17937 Review: https://reviewboard.asterisk.org/r/1356/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333689 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	64cb39333e	Add documentation for new manager event in chan_local AST-17623 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333071 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	901e275c4c	Add option for logging congested calls as CONGESTION instead of NO_ANSWER in CDR This patch adds a CDR option to cdr.conf that will allow CDR files to log calls ending with congestion in a way that is unique from other unanswered calls. (closes issue ASTERISK-14842) Reported by: Alec Davis Patches: cdr_congestion.diff.txt (License #5546) patch uploaded by Alec Davis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332760 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	3ad6dccac8	Merged revisions 332101 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332101 \| rmudgett \| 2011-08-16 12:17:28 -0500 (Tue, 16 Aug 2011) \| 140 lines Merged revisions 332100 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332100 \| rmudgett \| 2011-08-16 11:31:36 -0500 (Tue, 16 Aug 2011) \| 133 lines Fix multiple parking issues. JIRA ASTERISK-17183 Multi-parkinglot directs calls to wrong parkinglot. JIRA ASTERISK-17870 Cannot retrieve parked calls. JIRA ASTERISK-17430 ParkedCall() with no extension should pickup first available call and does not. JIRA AST-576 Issues with parking lots * Removed searching for parking lots by extension. Parking lots can only be found by the parking lot name since parking lot access extensions and spaces are not guaranteed to be unique. * Added parking_lot_name option to the Park and ParkedCall applications. Updated documentation for Park and ParkedCall applications. * Add parkext_exclusive configuration option to make parking entry extensions specify which parking lot they access. (closes issue ASTERISK-17183) Reported by: David Cabrejos Tested by: rmudgett, David Cabrejos (closes issue ASTERISK-17870) Reported by: Remi Quezada (closes issue ASTERISK-17430) Reported by: Philippe Lindheimer JIRA ASTERISK-17452 Parking_offset not used JIRA AST-624 'next' setting for findslot does nothing * Reimplemented since findslot feature option broken by -r114655. (closes issue ASTERISK-17452) Reported by: David Woolley Tested by: rmudgett JIRA ASTERISK-15792 Dialplan continues execution after transfer to park. This happens for DTMF attended transfer, DTMF blind transfer, and DTMF one-touch-parking if the party initiating these features also initiated the call. * Fixed the return code from the affected builtin features when parking a call. (closes issue ASTERISK-15792) Reported by: Mat Murdock Tested by: rmudgett, twilson JIRA AST-607 The courtesytone is not playing to the expected call when picking up a parked call. This is mostly a documentation problem. However, the option is not reset to the default when features.conf is reloaded. * Updated features.conf.sample documentation for courtesytone and parkedplay options. * Reset the parkedplay option to default when features.conf is reloaded. JIRA AST-615 AMI Park action followed by features reload results in orphaned channels in parking lot. * Reloading features.conf will not touch parking lots that have calls still parked in them. Reload again at a later time. Misc additional fixes: * Added unit test for parking lot dialplan usage checking. * Made update connected line when a parked call is retrieved from a parking lot. * Made retrieved parked call stop ringing or MOH depending upon how the call was waiting in the parking lot. * Made CLI "features show" indicate if the parking lot is enabled for use. * Added PARKINGDYNEXTEN channel variable to allow dynamic parking lots to specify the parking lot access extension. * Made AMI ParkedCalls action ParkedCall events have a Parkinglot header. * Made AMI ParkedCalls action ParkedCallsComplete event have a Total header. * Fixed potential deadlock from AMI Park action holding channel locks while calling masq_park_call(). * Fixed several places where ast_strdupa() were used inside of loops. (Mostly fixed by refactoring the loop body into its own function.) * Fixed copy_parkinglot() copying too much from the source parking lot. Extracted the parking lot configuration settings into struct parkinglot_cfg. * Refactored courtesytone playing code to put the channel not playing the tone in autoservice. * Fix when pbx-parkingfailed is played that the other channel is put in autoservice if it exists. * Fixed parkinglot reference leak in parked_call_exec() error paths. * Fixed parkinglot_unref() use of parkinglot after it was unreffed. * Made destroy the struct ast_parkinglot parkings lock when done. * Refactored the features.conf parking lot configuration code to eliminate redundancy. * Fixed feature reload to better protect parking lots. * Fixed parking lot container reference leak in handle_parkedcalls(). * Fixed the total count in handle_parkedcalls(). Review: https://reviewboard.asterisk.org/r/1358/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332117 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago

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